chromium/third_party/webrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler_unittest.cc

/*
 *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */
#include "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h"

#include <tuple>

#include "modules/audio_processing/test/audio_buffer_tools.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"

namespace webrtc {
namespace {

float SampleValueForChannel(int channel) { … }

void PopulateBuffer(AudioBuffer& audio_buffer) { … }

constexpr int kNumFramesToProcess = …;

class AudioSamplesScalerTest
    : public ::testing::Test,
      public ::testing::WithParamInterface<std::tuple<int, int, float>> { … };

INSTANTIATE_TEST_SUITE_P(…);

TEST_P(AudioSamplesScalerTest, InitialGainIsRespected) { … }

TEST_P(AudioSamplesScalerTest, VerifyGainAdjustment) { … }

TEST(AudioSamplesScaler, UpwardsClamping) { … }

TEST(AudioSamplesScaler, DownwardsClamping) { … }

}  // namespace
}  // namespace webrtc